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Telecommunications

Telecommunication occurs when the exchange of information between two or more entities (communication) includes the use of technology. Communication technology uses channels to transmit information (as electrical signals), either over a physical medium (such as signal cables), or in the form of electromagnetic waves. The word is often used in its plural form, telecommunications, because it involves many different technologies.

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Hi All,

We currently have an issue with a new build at a remote site.

The overall voice network is fully working at other locations, however the new site is having issues with inbound calls from the PSTN. The phones at both ends (internal and external) will ring, however no audio is passed. The call remains open, but silent.

Calls work outbound from the site successfully. The CUCM/Cube are on the main site, where calls work fine. The remote site is connected to the main network over a site to site VPN.

The only difference between this and other sites is the allocated IP range. The Cisco phones on the remote site are all using public IP addresses, where the main network and other remote sites are utilising private address space.

Any thoughts or suggestions would be greatly recieved.

Many Thanks,

John
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Hi

Does anyone know what the state is with the Shoretel (mitel) director licences?  Particularly with extensions?  We need to add more and cannot find out much info about them.
Awaiting comms co to update me.
Same question regarding the phones.  I believe 212k are EoL.  We need the line keys so what options do we have?

Thanks
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NEC SL2100 phone system started behaving poorly a week ago Sunday.  Rebooted all network equipment, and nothing changed.  Reboot the phone system and everything worked for about 20 minutes.  Same cycle of events, about same time frame.  Next checked all network equipment for firmware upgrades.  Wireless bridge Engenius EnStationAC went from 2.x to 3.1.2.11 on both sides successfully.  Problem returned yet again, but now a delay in getting dial tone of about 3-10 seconds.  If phone hangs up and connects immediately, no delay happens.  A monitoring computer is the only non-phone device on the entire voice network.  It shows random packet loss to phones.  The phone problem was originally they begain restarting at random.  They are not all happening at one time.  The delay never happens to VOIP phone on the system side of wireless bridge.  Engenius has yet to comment on the issue.  One phone (Only one phone) peforms IGMP actions accoring to Wire Shark.  Removing that specific phone solved the problem for almost 2 hours.  Yet it eventually returned.  All the time I have brought another laptop in for testing and moving it to both sides of the bridge.  If I am on the system side, I perform ping tests to the phones on the far side of bridge.  If I am testing the phone system resources, I am on the opposite side of bridge.  I perform around 1,500 packets to 3 devices simultaneously each time I test.  No latency or packet loss ever.  I have introduced a new PoE switch on the system …
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Dear Experts, is this diagram correct?

voip.PNG
We have 500 users, intend to use both VoIP service and Analog Tel service for backup. THere are 17 analog numbers to keep, so we choose Gateway GSM1024; for VoIP we use Grandstream UCM6510. We have several questions, can you please suggest?

- Can we use both UCM6510 and GSM1024 at the same time?
- Can we configure so that 1 department will use Analog line (GSM1024); other departments will use VoIP (UCM6510)?
- If one of these 2 fails , can the IP phone automatically change to the other, so that telephone service will NOT be interrupted?  
- Where can we find the reference link and installation manual for these devices?

Many thanks!
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I have a data jack that is terminated; but, not properly labelled.  I am trying to use a tone generator to find the correct patch panel connection that I need to use and cross connect the ethernet correctly to the end point.

https://www.youtube.com/watch?v=JHRIiw3OR6Y

The problem is that this specific data jack is in a machine shop floor and when I activate the toner every patch panel connection is triggered in the room.  If I turn off the toner generator every patch panel is still sounding off in that room.

If I try the toner generator in different locations(other rooms) there is no problem.  The toner generator works as I expect it to.

What could be causing the toner tester to sound off in that 1 specific patch panel?  Could it be that the ethernet cables were not grounded correctly?
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Hello,
I have about 150 CISCO 7960 phones and 50 CISCO 7942 PHONES, I AM SEEKING HELP GETTING THE 7942 PHONES UP AND RUNNING ON MY FREEPBX.... MY SERVER IS IN THE CLOUD... ANY HELP WOULD BE GREAT AND ILL PAY A FAIR PRICE TO MAKE THIS HAPPEN

RANDY
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Line is not releasing after transfer the call form Avaya SIP extension which is registered with Alcatel PABX
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In Skype for Business Server 2015, Is there a way to disable clients from retrieving location information from the LIS/Secondary LIS every time the client registers with the server? I want to prevent the client from performing a HELD request to get location information when it first registers to the server. Is this possible with the on-prem server?
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Lately, our staff complain about receiving multiple spam calls daily from what appear to be 'spoofed' local numbers. It started about two weeks ago and it looks like somehow direct numbers of staff got in the hands of wrong people.

Is there something that can be done about this?


Thank you!
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Ok, this is the last part to getting our site paging worked out.
I need to know how I can get our ShoreTel phones speakerphones to work with Valcom speakers when a page is sent out.
We have a ShoreTel system 14.2 director and Valcom v-2003A paging controller.  We need our system to page all phones and all Valcom speakers at the same time.
Can this be accomplished?
If this scenario cannot be accomplished with our current equipment then what would I need to replace/upgrade?
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Hello everyone,
We have a ShoreTel VoIP phone system and we would like to use it to page different zones from the ShoreTel desk sets.
We have a 3 zone Valcom Page Control Unit.
Our paging goal is this:
  1. Page outside only
  2. Page inside only
  3. Page outside and inside at the same time
Is there a solution to accomplish these scenarios?
Do we replace our current Valcom 3 zone paging control unit with a new paging control unit capable of accomplishing the 3 scenarios?
Thanks
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Hi all, we need to have call recording for our VOIP system, does any know of a 3rd party vendor who offers this? Our current vendor and we are not switching anytime soon.
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i am wondering if anyone has any experience in telephone  call accounting software.  i have a client who uses a piece of software called phonecontrol (phonecontrol.com).
i am wondering if there are other options out there.  they want to be able to see call costs for extensions on their samsung phone system so they can on charge their tennants.  

the phone control version they have can no longer be activated after  we had to reinstall their windows OS.  and the software support has not been helpful.  so looking for alternatives.
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I have installed Google voice on my Sprint PCS iPhone X.

Now how can I make my Google Voice number replace my Sprint PCS iPhone X caller ID number so that everytime that I call someone my Google Voice (and not my regular Sprint PCS phone number) will display on all outgoing calls?

Where do I go to enable this setting?

I received the email below on June 1, 2018 and I'm not sure how this affects my Sprint PCS Google voice service:

We are contacting you to let you know that per previous communication, due to upcoming upgrades to Sprint’s network, Sprint will no longer be supporting the Google Voice with Sprint integration. Effective today your Google Voice integration with your Sprint phone number (xxx) xxx-xxxx has been disabled by Sprint.

Effective today:
All outgoing calls (including international calls) and texts will be made through Sprint at Sprint’s calling and texting rates, if applicable.
All new messages, calls, and voicemails sent from your Sprint phone will not be stored in Google Voice. You will still be able to see your messages, voicemail, and call history from before June 1, 2018 in Google Voice on your Sprint device. You can also export this data from your Google Voice account at takeout.google.com.
You won’t be able to use Google Voice-enabled capabilities such as call forwarding, voicemail transcription, spam detection, and other Google Voice features. These capabilities can be enabled from your Sprint device. Click here for more information …
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android 7
automatic call recorder
free version
app version 5.42.1

recordings were good then I noticed last recording was at a low volume
Maybe in a version upgrade
volume is lower which is very difficult to hear
If I pay for paid version, do you think this will fix.
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I have a remote extension (yealink handset) that is dropping outbound calls after 35 seconds. I have diagnosed through logging that the asterisk server is not receiving the proper ACK, and the reason for that is that the asterisk server is looking for the reply from 192.168.1.4, which is the local subnet of the remote phone. Obviously the asterisk server will never receive a reply from that address. What changes do I need to make so that the asterisk server is looking for replies from the WAN IP of the remote network? NAT is setup properly on both ends. For reference, this is CompletePBX from Xorcom that we're dealing with and a copy of the asterisk log error is below.

[2018-08-16 09:27:08] WARNING[5720] chan_sip.c: Retransmission timeout reached on transmission 405286438@192.168.1.4 for seqno 2 (Critical Response)
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A customer of mine is moving their people from Office A to Office B, but is leaving the servers at Office A.

There are 5 networks in 3 different security zones that need to exist at Office B for the users, but the companies procurement department has vetoed using a Layer 2 type service because it's not in the current contract with their Telco provider.  :/

Net 1 - 100mb - Security Zone 1
Net 2 - 100mb - Security  Zone 1
Net 3 - 100mb -  Security Zone 2
Net 4 - 10mb -  Security Zone 3
Net 5 - 10mb - Security  Zone 3

The nets are all class C.  All the systems at Office B are new, and having different IP addresses is (amazingly / apparently) not a problem.

The Telco provider has sold them a pair of diverse 300 MB Ethernet MPLS circuits with three private VPNs which will be connected to two new Cisco 4431's at Office A and Office B that will both connect to all 5 network switches at Office A.  These routers will not directly route any packets between the 5 nets / 3 zones (there are other existing routers and firewalls at Office A that will do those tasks if needed), they are purely for connectivity between Office A and Office B.

Within the order notes for the circuits, I see the wording "Each MPLS CE router will utilize the Multi-VRF feature to segment traffic by application.  A total of 3 VPN's will be utilized."

I have no problems with routing protocol features (EIGRP or BGP) on high availability networks, and although I have never configured it …
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I am converting to SIP at a company site.  I set up the fortigate 60d identical to another one of our company sites where SIP is already implemented.

The SIP Cloud provider tells me that the firewall is not "passing confirmation packets  the phone system (PBX) sends out when it detects an incoming call",
and this is causing their system to transfer the call to our failover number.

I have opened up all requested ports (TCP/UDP etc), configurd policy,and QOS to high priority- everything they suggested, and as I mentioned earlier, it is configured exactly like our other site's fw.

I suspect it is a configuration with one of their servers.  In any case, Logging for all events is enabled in the firewall policy.  How can I tell if the firewall is blocking/not passing back the confirmation packets they SIp provider mentions?
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We have been using Skype for Business (formerly Lync) for IM and video conferencing but are thinking about using Skype PSTN as a replacement for our current PBX.

We are in the process of organising a trial for some users to pilot Skype for Business as their only phone option.

One of the things we need to do is gather some data once their trial has finished and I am looking for help identifying test criteria for the new system.

If you have any thoughts about the sort of questions we should be asking users to assess the suitability of Skype for Business, then please let me know.

Thoughts so far:-

Equipment
Which microphone device do you normally use for S4B phone calls?
Which amplification device do you normally use for S4B phone calls?
Frequency of use
In a typical day, how likely are you to make a phonecall using S4B
In a typical day, how likely are you to receive a phonecall using S4B
Sound Quality
How does the sound quality compare to landline
How does it compare to mobile

Any questions / comments appreciated.


Jon
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When I call my cell phone using skype on my windows 10 computer
I see the caller id that I selected on skype.com
I am spoofing my phone number to display another phone number that I use but am not currently using because I am calling from a computer.

But when I call a big business the number that shows up is not mine and can not be called again.

How do I purchase the caller id system that a big business has.
Caller ID that does not work with a spoofed phone number.
1
good day,

I have two switch both configured for vlan 869 (Voice). The phones connected to SW01 can get IP address and phones connected to SW02 cannot get IP address. Can someone assist where I made configuration mistake. This is my first config as I am learning the Cisco commands. Attached are my configurations for both switch.
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Hi, we have a data network here with Cisco switches that we manage. Now, there is a also a VOIP vendor who his own switches in our network. (He didnt want to use ours and VLAN everything).

Both the networks are on different subnets. Now, we noticed all of a sudden PC's started getting IP's from the phone subnet...and it's wreaking havoc internally. I tried to manually trace all 200+ cables in the office to see if someone plugged a phone device into the data network, but no luck..

How else can i troubleshoot this from say a switch level?
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For people that own / manage 800 numbers for businesses:

Do you get charged for / pay for calls from payphones?  Years ago there was a charge - 26c I think it was - that people that own toll free numbers were charged that went to the pay phone owner.  Wonder if that's still in effect. And for bigger businesses, do they (the 800 number supplier) not charge the 800 number owner?

And 800 numbers - does the owner typically pay a per minute cost? Or has it moved like outgoing phone lines to a flat fee / month?

thanks!
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Users are being forced to upgrade to latest skype client, currently v8.25, but from what i can see the user is not able to increase the chat text size.   This is troublesome and a backwards step for particularity elderly people using 1080+ screens.  

Skype (version 7) for Windows will stop working soon. Update to the new Skype (version 8) for Windows today! Don’t miss out on exceptional HD video calling, 300 MB file transfers, the ability to tag contacts with @‍mentions and more. Upgrade to the latest version of Skype to seamlessly transfer your sign in information, contacts and chat history. For more details and help, visit this page.

Does anyone have a hack/workaround for the text size?   If Microsoft have not built the setting into the initial design I can't imagine they will be adding it anytime soon.

Thanks,
--Paul
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help with setting up VLAN on a few switches for phone and data equipment.

i have a series of managed switches that are uplinked together.  I would like to set up a VLAN 100 for a dedicated router that is on port 48 of one switch. This router will listen to requests from phones that are plugged into any other random ports on the switch.  This switch is a ubiquiti unit that allows me to set port 48 to listen to vlan 100 traffic only.

The phones are set to 802.1Q with a vlan of 100.  there are other computers and servers on the switch that are on a 192.168.0.x subnet.  The server is handing out DHCP as well as the router on port 48.  The idea is to isolate the traffic for the phones to ONLY communicate with the DHCP server on port 48.  

Right now, this setting is working. However my question to you, is since the phones are all plugged into random ports 1-47 and set to vlan100  and these ports are set to listen to both default lan traffic as well as vlan100...am i simply congesting the switch with added default and vlan traffic vs setting the actual ports that the phones are plugged into to ONLY vlan 100?

Also, if i plug in another switch,, do i need to set the uplink from one switch to another switch with a vlan100 for them to comminicate or will they pass the phones traffic that is tagged 802.1Q VLAN 100 traffic to the other where the port 48 will ultimately listen and grab it? Thank you!
0

Telecommunications

Telecommunication occurs when the exchange of information between two or more entities (communication) includes the use of technology. Communication technology uses channels to transmit information (as electrical signals), either over a physical medium (such as signal cables), or in the form of electromagnetic waves. The word is often used in its plural form, telecommunications, because it involves many different technologies.